Conventionally, voice data remains separate from network data traffic. In particular, many enterprises will have a data network connecting a variety of workstations as well as a public branch exchange (PBX) for voice data. As data networks proliferate, it is becoming an increasingly desirable goal to integrate transmission of voice and data.
Transmitting data over voice systems can be inefficient, as such systems typically transmit voice and data in predetermined frames. Frames are allocated portions of bandwidth, usually formed according to a time division multiplexing scheme. In the event voice or data traffic drops, a frame may be transmitted with only voice or data, thus wasting available bandwidth.
Transmitting voice over a data network (voice over network) can provide advantages over other approaches. Voice over network can take advantage of existing network structures. Further, as noted above, networks (including the Internet) continue to grow in size and in bandwidth. Voice over network can be more efficient than data over voice as such systems are typically packet based. In a packet based system, bandwidth is used as needed. When data is transmitted, a data packet is sent through the network. When voice is needed, a voice packet is transmitted through the network.
Voice over networks can provide additional cost savings as voice may be transmitted between location without incurring conventional toll charges. This can be particularly true for systems that transmit voice over the Internet.
Many networks, can be “connectionless” networks. Connectionless networks can provide multiple possible paths between source and destination. Consequently, in some cases, voice transmitted over such networks may be more reliable as voice data can reach a destination even if some of the network nodes are not operational.
Voice over data networks may provide additional features in a cost-effective fashion. In a particular, transmitting voice over a data network may allow for “multicasting” of voice data (transmission of voice data to multiple destinations) and/or mixed media transmissions (voice and data) as but one example.
One type of voice of network approach utilizes the Internet protocol (IP), and is often referred to as voice-over-IP (VoIP).
Data networks may take a variety of forms. As noted above, a data network may be a connectionless network, including the Internet. Further, a network may include portions that overlaid and/or integrate connection-oriented legs. Such systems include internet protocol (IP) over asynchronous transfer mode (ATM), IP switching systems, and multiprotocol label switching (MPLS) or similar switching systems.
Various proposals for implementing voice over data networks have been proposed. One general approach is the idea of a network “gateway.” A network gateway can provide access to a network (such as the Internet) for a variety of conventional voice data sources (voice channels). As but one example, a network gateway can be an IP gateway that integrates a PBX with an IP network. In such an arrangement, users may make telephone calls that appear entirely conventional, but are in fact being transmitted over a data network.
One drawback associated with voice over data networks can be latency. Latency is the delay introduced by the system into a voice transmission. Various sources may contribute to latency. A transmitting source introduces some delay in placing the voice into packet form. Typically the voice data can be digitized and/or compressed and then placed in packet form. Transmission of the voice over a data network can also introduce latency. Routing from node to node, or along a switching path, can consume additional time. Finally, a receiving destination can introduce delay. Voice data can be extracted from a packet and then transmitted along an appropriate voice channel. Such packet processing can be conventionally accomplished with a general purpose processor.
According to one conventional approach, a network voice packet can be forwarded to a destination according to conventional protocols, such as TCP/IP. A destination can include conventional network hardware, such as a network interface card, which can receive the packet. A general purpose processor can then extract voice data within the packet (including any decoding/decompressing) and then play the voice data over sound card, or the like. Such computation steps can contribute significantly to latency.
According to another conventional approach, a network can include a network phone (such as an “IP Phone”). A network phone can have its own network address, and include packet processing and voice data processing hardware.
One conventional way to decrease latency is with more complex routers. In particular, some routers may be capable of giving particular packets precedence over others. As but one example, some packet headers can include a “type of service” or “quality of service” field. A network voice packet can be given higher precedence than other packets, thereby reducing latency when compared to other packets. A drawback to such approaches is that all routers/switches in the network must have such an advanced processing capability, and thus can be expensive to implement. Consequently, such an approach may be better suited for enterprise wide area networks (WANs), instead of a larger network structure such as the Internet.
Another approach can essentially reserve a route for a voice packet flow. One such approach is the Resource ReserVation Protocol (RSVP). RSVP, like the precedence approach noted above, can rely on more complex routers and thus be an expensive solution.
While the approaches described above can reduce latency for voice packet data, such approaches may be difficult and/or expensive to implement. Further, it is still desirable to reduce latency even more. Thus, further ways of decreasing latency in network voice packets could a valuable contribution to voice over data networking.
While conventional packet forwarding of voice a data network may provide adequate results, this may not always be the case. To help improve the processing of time dependent data, including voice data, a number of protocols have been proposed. One such protocol is the Real-Time Transport Protocol (RTP). An RTP header can provide sequence and timestamp information than may help assemble voice data once it has been received.
Another concern that may arise out of voice over network systems is different possible voice data formats. Some formats may require different processing than others. Further, the processing of one voice format type may take longer and/or require more resources than other voice formats. Different voice formats may include, as but two of the many possible examples, “multiplexed” voice packets, which can contain voice data from multiple sources, and “simplex” voice packets, which may contain voice data form a single source. With the advent of multiplexed and simplex voice packets, systems may have to be capable of accommodating both types of voice packets.
The need for processing both simplex and voice data packets may further contribute to overall latency between a voice source and the resulting audio destination. Further, the processing of simplex and multiplexed voice data packets may add complexity to the instructions in conventional systems that store the packet to memory and then deprocess the packet according to instructions.
It would be desirable to arrive at a system that can improve the speed and efficiency at which simplex and multiplex voice data packets are processed.